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WavPack compression (.WV files) can compress (and restore) 8-, 16-, 24-, and 32-bit fixed-point, and 32-bit floating point audio files in the .WAV file format. It also supports surround sound streams and high frequency sampling rates. Like other lossless compression schemes, the data reduction rate varies with the source, but it is generally between 30% and 70% for typical popular music and somewhat better than that for classical music and other sources with greater dynamic range.[1]

WavPack also incorporates a "hybrid" mode which still provides the features of lossless compression, but it creates two files: a relatively small, high-quality, lossy file (.wv) that can be used by itself; and a "correction" file (.wvc) that, when combined with the lossy file, provides full lossless restoration. This allows the use of lossy and lossless codecs together.

David Bryant started development on WavPack in mid-1998 with the release of version 1.0 (1998-08-15). This first version compressed and decompressed audio losslessly, and it already featured one of the best efficiency vs. speed ratios among lossless encoders.[2]

Very soon after the release of version 1.0, v. 2.0 (2 September 1998) was released, featuring lossy encoding (using only quantization of prediction residue for data reduction - no psychoacoustic masking model was applied to the stream).

In 1999, version 3.0 (12 September 1999) was released, with a new "fast mode" (albeit with reduced compression ratio), compression of raw (headerless) PCM audio files, and error detection using a 32-bit cyclic redundancy check.

WavPack development is ongoing. A feature added in late 3.x versions is the "hybrid" mode where the encoder generates a lossy file and a correction file such that both can be decompressed back to a PCM stream that is same quality as the original. A “roadmap” is also published by the author, containing possible hints on future development.[3]

To ensure high-speed operation, WavPack uses a predictor that is implemented entirely in integer math.[6] In its "fast" mode the prediction is simply the arithmetic extrapolation of the previous two samples. For example, if the previous two samples were −10 and 20, then the prediction would be 50. For the default mode a simple adaptive factor is added to weigh the influence of the earlier sample on the prediction. In our example the resulting prediction could then vary between 20 for no influence to 50 for full influence. This weight factor is constantly updated based on the audio data's changing spectral characteristics.

The prediction generated is then subtracted from the actual sample to be encoded to generate the error value. In mono mode this value is sent directly to the coder. However, stereo signals tend to have some correlation between the two channels that can be further exploited. Therefore, two error values are calculated that represent the difference and average of the left and right error values. In the "fast" mode of operation these two new values are simply sent to the coder instead of the left and right values. In the default mode, the difference value is always sent to the coder along with one of the other three values (average, left, or right). An adaptive algorithm continuously determines the most efficient of the three to send based on the changing balance of the channels.

Instead of Rice coding, a special data encoder for WavPack is used. Rice coding is the optimal bit coding for this type of data, and WavPack's encoder is less efficient, but only by about 0.15 bits/sample (or less than 1% for 16-bit data). However, there are some advantages in exchange; the first one is that WavPack's encoder does not require the data to be buffered ahead of encoding; instead it converts each sample directly to bitcodes. This is more computationally efficient, and it is better in some applications where coding delay is critical. The second advantage is that it is easily adaptable to lossy encoding, since all significant bits (except the implied "one" MSB) are transmitted directly. In this way it is possible to only transmit, for example, the 3 most significant bits (with sign) of each sample. In fact, it is possible to transmit only the sign and implied MSB for each sample with an average of only 3.65 bits/sample.

This coding scheme is used to implement the "lossy" mode of WavPack. In the "fast" mode the output of the non-adaptive decorrelator is simply rounded to the nearest codable value for the specified number of bits. In the default mode the adaptive decorrelator is used (which reduces the average noise about 1 dB) and both the current and the next sample are considered in choosing the better of the two available codes (which reduces noise another 1 dB).

No floating-point arithmetic is used in WavPack's data path because, according to the author, integer operations are less susceptible to subtle chip-to-chip variations that could corrupt the lossless nature of the compression (the Pentium floating point bug being an example). It is possible that a lossless compressor that used floating-point math could generate different output when running on that faulty Pentium. Even disregarding actual bugs, floating-point math is complicated enough that there could be subtle differences between "correct" implementations that could cause trouble for this type of application.[7] A 32-bit error detection code to the generated streams is included to maintain user confidence in the integrity of WavPack's compression.

A cut-down version of WavPack was developed for the Texas Instruments TMS320 series Digital Signal Processor. This was aimed predominantly at encouraging manufacturers to incorporate WavPack compression (and de-compression) into portable memory audio recorders. This version supported features that were applicable only to embedded applications (stream compression in real-time, selectable compression rate) and dropped off features that only applied to full computer systems (self extraction, high compression modes, 32-bit floats). The TMS320 series DSPs are native integer devices, and support WavPack well. Some 'special' features of the full WavPack software were included (ability to generate a correction 'file' (stream) for example) and others were excluded. The port was based on version 4.

WavPack support was added to WinZip starting with version 11.0 beta, released in October 2006.[8] This extension to the ZIP file format was included by PKWARE, the maintainers of the format, in the official APPNOTE.TXT description file starting with version 6.3.2, released on 28 September 2007.[9]